commit | c335b0e63bff56ca0fbfa617dee6a644c85df164 | [log] [tgz] |
---|---|---|
author | Henrik Boström <hbos@webrtc.org> | Thu Apr 08 05:25:38 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Apr 08 06:39:22 2021 |
tree | f45d2b762571dc61f17cc0380000b87955fc9201 | |
parent | 3e0c60ba4ef28a9f26fe991e5eec3150402c7dd3 [diff] |
[Unified Plan] Don't end audio tracks when SSRC changes. The RemoteAudioSource has an AudioDataProxy that acts as a sink, passing along data from AudioRecvStreams to the RemoteAudioSource. If an SSRC is changed (or other reconfiguration happens) with SDP, the recv stream and proxy get recreated. In Plan B, because remote tracks maps 1:1 with SSRCs, it made sense to end remote track/audio source in response to this. In Plan B, a new receiver, with a new track and a new proxy would be created for the new SSRC. In Unified Plan however, remote tracks correspond to m= sections. The remote track should only end on port:0 (or RTCP BYE or timeout, etc), not because the recv stream of an m= section is recreated. The code already supports changing SSRC and this is working correctly, but because ~AudioDataProxy() would end the source this would cause the MediaStreamTrack of the receiver to end (even though the media engine is still processing the remote audio stream correctly under the hood). This issue only happened on audio tracks, and because of timing of PostTasks the track would kEnd in Chromium *after* promise.then(). This CL fixes that issue by not ending the source when the proxy is destroyed. Destroying a recv stream is a temporary action in Unified Plan, unless stopped. Tests are added ensuring tracks are kLive. I have manually verified that this CL fixes the issue and that both audio and video is flowing through the entire pipeline: https://jsfiddle.net/henbos/h21xec97/122/ Bug: chromium:1121454 Change-Id: Ic21ac8ea263ccf021b96a14d3e4e3b24eb756c86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214136 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33645}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.